Log - 2ae140ae7e34bd1e102ca832a14b7b0adbb72d15 - platform/external/webrtc

57858b3 Reland "Update RTCStatsCollector to work with RtpTransceivers" by Steve Anton · 7 years ago
ee2388f Revert "Update RTCStatsCollector to work with RtpTransceivers" by Guido Urdaneta · 7 years ago
56bae8d Update RTCStatsCollector to work with RtpTransceivers by Steve Anton · 7 years ago
5dfde18 Change PeerConnection stats interface to be more flexible by Steve Anton · 7 years ago
76d2952 Don't crash when sender info has been discarded by lower layers. by Harald Alvestrand · 7 years ago
2d8609c Move internal PeerConnection methods to PeerConnectionInternal by Steve Anton · 7 years ago
b8e1201 Generate track stats when SSRC=0 by Harald Alvestrand · 7 years ago
a3dab84 Refactor stream stats generation by Harald Alvestrand · 7 years ago
c72af93 Reland "Move stats ID generation from SSRC to local ID" by Harald Alvestrand · 7 years ago
c0092c3 Revert "Move stats ID generation from SSRC to local ID" by Erik Språng · 7 years ago
e357a4d Move stats ID generation from SSRC to local ID by Harald Alvestrand · 7 years ago
8906187 Pivot generation of stats to iterate senders/receivers by Harald Alvestrand · 7 years ago
593e325 Change RTCStatsCollector to only access channels from signaling thread by Steve Anton · 7 years ago
719487e Generate signed packets_lost in WebRTC-stats by Harald Alvestrand · 7 years ago
56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
37e489c Add network_type to local RTCIceCandidateStats by Gary Liu · 7 years ago
89e7126 Optional: Use nullopt and implicit construction in /pc/rtcstatscollector.cc by Oskar Sundbom · 7 years ago
36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
d5585ca Move almost all references from WebRtcSession to PeerConnection by Steve Anton · 7 years ago
e2d6a06 Reland "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
1af3d82 Revert "Reland "Clean up libjingle API dependencies."" by Henrik Kjellander · 7 years ago
9185aca Reland "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
978b876 Move clients of WebRtcSession to use PeerConnection by Steve Anton · 7 years ago
b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
581df61 Revert "Reland "Clean up libjingle API dependencies."" by Patrik Höglund · 7 years ago
5117b04 Reland "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
7bcfc3b Revert "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
bf66794 Revert "Move clients of WebRtcSession to use PeerConnection" by Alex Loiko · 7 years ago
57fb315 Clean up libjingle API dependencies. by Patrik Höglund · 7 years ago
3dc4d4a Move clients of WebRtcSession to use PeerConnection by Steve Anton · 7 years ago
9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/rtcstatscollector.cc]
8ab0fd8 Reland of Trace the stats report as JSON instead of each stat separately. (patchset #1 id:1 of https://codereview.webrtc.org/3001683002/ ) by ehmaldonado · 7 years ago
2dbc69f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 7 years ago
3439c89 Revert of Trace the stats report as JSON instead of each stat separately. (patchset #3 id:100001 of https://codereview.webrtc.org/2986453002/ ) by mbonadei · 7 years ago
80c6578 Trace the stats report as JSON instead of each stat separately. by ehmaldonado · 7 years ago
80c829f Enable tracing on rtcstats_integrationtest.cc by ehmaldonado · 7 years ago
b0b721a Increase the size of the buffer for type.name.id. by ehmaldonado · 7 years ago
a26196b Trace stats in RTCStatsCollector. by ehmaldonado · 7 years ago
e76bd3a Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy. by zstein · 7 years ago
7b0c6fa RTCStatsCollector: Get track IDs from senders/receivers instead of streams. by hbos · 7 years ago
c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
f79ade1 Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )" by stefan · 7 years ago
d72098a Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ ) by charujain · 7 years ago
e80f4c9 Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. by Stefan Holmer · 7 years ago
5bf9def RTCStatsCollector: Remove closed channels from opened set. by hbos · 7 years ago
a7a9be1 Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect. by hbos · 8 years ago
13f54b2 Rename RTCCodecStats.codec -> mimeType, parameters -> sdpFmtpLine. by hbos · 8 years ago
bf8d3e5 RTCIceCandidatePairStats.[total/current]RoundTripTime collected. by hbos · 8 years ago
92eaec6 RTCIceCandidatePairStats.nominated collected. by hbos · 8 years ago
a51d4f3 Re-land of RTCInboundRTPStreamStats.qpSum collected. by hbos · 8 years ago
ed02c6d Revert of RTCInboundRTPStreamStats.qpSum collected. (patchset #4 id:80001 of https://codereview.webrtc.org/2675943002/ ) by skvlad · 8 years ago
cd195be RTCInboundRTPStreamStats.qpSum collected. by hbos · 8 years ago
338f78a RTCIceCandidatePairStats.available[Outgoing/Incoming]Bitrate collected. by hbos · 8 years ago
3443bb7 RTCRTPStreamStats.ssrc changed type to uint32_t. by hbos · 8 years ago
585a9b1 Refactor and clean-up relating to RTCCodecStats. by hbos · 8 years ago
e702b30 Adding C++ versions of currently spec'd "RtpParameters" structs. by deadbeef · 8 years ago
b0ae920 RTCRTPStreamStats.mediaTrackId renamed to trackId. by hbos · 8 years ago
50cfe1f RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector. by hbos · 8 years ago
7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (99%) from webrtc/api/rtcstatscollector.cc]
f64941f RTCMediaStreamTrackStats.framesDecoded collected. by hbos · 8 years ago
fefe076 RTCMediaStreamTrackStats.framesSent collected by RTCStatsCollector. by hbos · 8 years ago
42f6d2f RTCMediaStreamTrackStats.framesReceived collected by RTCStatsCollector. by hbos · 8 years ago
9e30274 RTCMediaStreamTrackStats collected on a per-attachment basis. by hbos · 8 years ago
160e4a7 RTCMediaStreamTrackStats.kind added and collected. by hbos · 8 years ago
7064d59 RTCTransportStats.dtlsState replaces .activeConnection by hbos · 8 years ago
84abeb1 RTC[In/Out]boundRTPStreamStats.mediaTrackId collected. by hbos · 8 years ago
953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
c0dad89 Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ ) by deadbeef · 8 years ago
67b3bbe Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago
b4e426e RTCIceCandidateStats.transportId added and collected. by hbos · 8 years ago
6769c49 RTC[In/Out]boundRTPStreamStats: qpSum,framesDecoded,framesEncoded added. by hbos · 8 years ago
06495bc RTCIceCandidatePairStats.[state/priority] added, ConnectionInfo updated. by hbos · 8 years ago
f415f8a Removed RTCStatsCollector::ProducePartialResultsOnWorkerThread. by hbos · 8 years ago
02d2a92 RTCStatsReport::AddStats DCHECKs that the ID is unique. by hbos · 8 years ago
b78306a Fix segfault when PeerConnection is destroyed during stats collection. by hbos · 8 years ago
df6075a RTCStatsCollector: Utilize network thread to minimize thread hops. by hbos · 8 years ago
3168c7a Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time. by hbos · 8 years ago
e10e6d1 RTCOutboundRTPStreamStats.roundTripTime: Only report non-negative values. by hbos · 8 years ago
9a394f0 Skip RTCMediaStreamTrackStats.echoReturnLoss[Enhancement] default value. by hbos · 8 years ago
e448dd5 RTCIceCandidatePairStats.consentRequestsSent set by RTCStatsCollector by hbos · 8 years ago
02cd4d6 RTCInboundRTPStreamStats.packetsLost set by RTCStatsCollector. by hbos · 8 years ago
d82f512 RTCIceCandidatePairStats.requestsReceived defined by RTCStatsCollector. by hbos · 8 years ago
0583b28 Collecting RTCIceCandidatePairStats.transport_id and improved unittests. by hbos · 8 years ago
0adb828 RTCCodecStats[1] added. by hbos · 8 years ago
820f578 RTCInboundRTPStreamStats's [fir/pli/nack]_count are collected for video. by hbos · 8 years ago
82ebe02 Correct stats for RTCPeerConnectionStats.dataChannels[Opened/Closed]. by hbos · 8 years ago
1b35d4c Don't copy loop variables when we don't have to by kwiberg · 8 years ago
09bc128 RTCMediaStream[Track]Stats added. by hbos · 8 years ago
eeafe94 RTCInboundRTPStreamStats[1] added. by hbos · 8 years ago
6ded190 RTCOutboundRTPStreamStats[1] added. by hbos · 8 years ago
7eaa836 Revert of RTCOutboundRTPStreamStats added. (patchset #3 id:80001 of https://codereview.webrtc.org/2456463002/ ) by perkj · 8 years ago
4ed0750 Revert of RTCInboundRTPStreamStats added. (patchset #4 id:100001 of https://codereview.webrtc.org/2452043002/ ) by perkj · 8 years ago
0d7bf16 RTCInboundRTPStreamStats[1] added. by hbos · 8 years ago
69e9cb0 RTCOutboundRTPStreamStats[1] added. by hbos · 8 years ago
02ba211 Move RTCStatsCollector helper functions to anonymous namespace. by hbos · 8 years ago
5d79a7c rtcstats_objects.h updated with TODOs about stats not being collected by hbos · 8 years ago